In order to achieve access independence and to maintain a smooth interoperation with wired terminals across the Internet, an IP multimedia subsystem (IMS) core network as specified e.g. in the 3GPP (Third Generation Partnership Project) specification TS 23.228 has been developed to be conformant to IETF (Internet Engineering Task Force) “Internet Standards”. The IMS enables network operators of mobile or cellular networks to offer their subscribers multimedia services based on and built upon Internet applications, services and protocols. The intention is to develop such services by mobile network operators and other third party suppliers including those in the Internet space using the mechanisms provided by the Internet and the IMS. The IMS thus enables conversion of, and access to, voice, video, messaging, data and web-based technologies for wireless users, and combines the growth of the Internet with the growth in mobile communications. In IMS, the Session Initiation Protocol (SIP) is used as the main session control protocol between end user equipments and Call State Control Functions (CSCFs) located in the IMS. SIP enables network operators to provide new features for end users such as dialing with the use of SIP Uniform Resource Indicators (SIP URIs).
The 3GPP Release 6 standard will introduce interworking between the IMS and the CS domain. 3GPP will however only specify procedures to have speech calls placed between the CS domain and IMS. Interworking is provided with introduction of Media Gateway Control Function (MGCF) and IMS-Media Gateway (IMS-MGW). Interworking of other type of calls such as video telephony are currently out of scope.
In order to provide similar speech service with service capabilities offered by IMS over the IP, sufficient end-to-end Quality of Service (QoS) is required. In most cases the sufficient QoS can be achieved from the core network but QoS in radio access networks can be difficult to achieve. This is especially the case in networks which does not have UMTS radio access or GERAN/GPRS radio access with support of conversational traffic class.
Such situation is present e.g. in United States of America or in general, countries that are either not capable of or have not planned to provide 3G in the near future. Also use of enhanced GPRS (E-GPRS) has been analysed to be problematical in order to support voice over IP connections.
Sufficient QoS for voice over IP cannot be achieved from 2G or 2.5G radio networks if conversational traffic class is not supported by user equipment (UE), radio access network and core network. This will prevent use of IMS based services for voice calls, if nothing is done, in the Public Land Mobile Networks (PLMNs). Also the roaming of IMS subscriber is limited by this lack of functionality, because when end user is roaming in such PLMNs that do not support sufficient QoS, CS services need to be used instead.
IETF and 3GPP are working on a SIP conferencing service. The goal is to define how conferencing type of service can be established between 3GPP compliant SIP terminals. Simultaneously with this work, it is also studied how the interworking between 3GPP IMS and legacy circuit-switched (CS) core network domains can be achieved. A cellular network, i.e. a Public Land Mobile Network (PLMN) can be regarded as an extension of networks with CS domains and packet switched (PS) domains within a common numbering plan and a common routing plan. The PLMN infrastructure is logically divided into a core network (CN) and an access network (AN) infrastructure, while the CN infrastructure is logically divided into a CS domain, a PS domain and an IMS. The CS and PS domains differ by the way they support user traffic. These two domains are overlapping, i.e. they contain some common entities. A PLMN can implement only one domain or both domains. In particular, the CS domain refers to the set of all CN entities offering CS type of connections for user traffic as well as all the entities supporting the related signaling. A CS type of connection is a connection for which dedicated network resources are allocated at the connection establishment and released at the connection release. The PS domain refers to the set of all CN entities offering PS type of connections for user traffic as well as all the entities supporting the related signaling. A PS type of connection transports the user information using autonomous concatenation of bits called packets, wherein each packet can be routed independently from the previous one. The IMS domain comprises all CN elements for provision of IP multimedia services comprising audio, video, text, chat, etc. and a combination of them delivered over the PS domain.
So far, conferencing has been considered from IMS point of view where simultaneously communicating parties are IMS subscribers with IMS subscription. This scope enables the full end to end use of SIP between the participants. SIP is an application-layer control protocol for creating, modifying and terminating sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. In SIP-based conferencing, a conference-URI (Uniform Resource Indicator), e.g.:
SIP: conf—234@conference.operator.com
is used to establish a conference. A drawback of this solution is that it is completely based on IP and thus cannot be used with GSM technology.
The 3GPP Release 5 standard specifies a limited set of services, like call forwarding, call line identification, etc. 3GPP Release 6 will introduce more services, conference call being one of them. Conference call is currently under standardization in IETF, and the 3GPP will adopt the service from there to its own standards.
The Release 6 conference call service will bring several benefits compared to the existing service e.g. in GSM network. In order to provide similar features in CS mobile networks, enhancements to the existing GSM service are needed. In particular, the conventional GSM conference call service lacks many of the features that have been planned in the All-IP conference framework. These features can be seen as requirements that the enhanced service must fulfill and cover easy and fast creation of the conference call from mobile terminal, while each participant has to know whether she/he has been added to the conference, who else is in the same conference, if others join the conference, if others leave the conference during the session, for how long the conference session has been running, or when others have been joined.
In GSM, the user needs to call the members and add them to the conference one-by-one. This obviously takes time if the group is large. Furthermore, the creator of the conference needs to set up a private call to the new participant, while the conference session is on hold. In this private call the creator usually tells the participant that he/she is being added to a conference. GSM does not provide an automatic mechanism to share this information. Usually the creator introduces all participants to each other, and tells to others that he/she is adding a new participant to the conference. The participant who is leaving tells it before doing so. A resulting problem is that the call might be dropped accidentally or by network or radio link failure. In these cases it might take a long time before the other participants realize this. The only way for a participant to know how long the session has been running or when other have been joined is to ask it from the creator or certain participants. Furthermore, the only way to moderate the conference is to tell orders in the voice channel. The only way to know who has a right to speak is to listen to the orders the moderator is telling. GSM does not provide an automatic mechanism to share the above mentioned different kinds of information. Furthermore, voice channels are used inefficiently since there is no central point in the network to mix the voice channels.
Moreover, if the conference creator drops out, the whole conference will be lost. In addition, the maximum number of participants in the GSM conference call, i.e. multiparty call, is six.